3 Way Calling
Accidental Disconnect Protection
Answer Groups/Keys
API -
• C
• Perl
• Python
• XML Via Perl
• Shell scripting
Autodial Keys 7960: yes
Automated Attendant -
• Custom Call Routing
• System Answer
Automatic Line Selection
Automatic Set Relocation Assumes use of DHCP
Auto Attendant
Auxiliary ringing Use ATA-186 or other adapter to connect to additional analog ringers
Background Music mp3 streaming or files standard. Line-in audio level 1 effort
Busy Lamp Indication NA Phones can only display local number status
Button Inquiry Read the rest of this entry »
The first 802.11 phones to hit the “popular” commercial market were introduced by Jeff Pulver’s company, Pulverinnovations.com a few weeks ago at the VON 2003 Fall show. While there are other 802.11 phones, this is the first one runs SIP that I have seen that is actually available and taking orders. The Cisco 7920 doesn’t support SIP yet, but rumors have it working by the end of the year (though the $500-$600 pricetag will discourage many buyers.)
The Manager interface is used to remotely view/change console (and other) actions. Through the use of the manager interface, you are able to send actions and receive responses very similar to the CLI interface. However Asterisklatest version 1.4 beta has GUI manager interface which is very user friendly. Asterisk comes with AsteriskManager API, which allows third-party software to provide external management services. And through the freely available Asterisk-Sounds add-on package, Asterisk provides a full range of canned voice messages to use with menus and IVR.
Asterisk Voicemail is called from within the dialplan (see /etc/asterisk/extensions.conf)like any other application. It takes as an argument the mailbox to be used and the message to be layed (busy or unavailable.) The message is set in the dialplan by preceding the mailbox number with ‘b’ or ‘u’ for busy or unavailable. Alternately, the mailbox number may be preceeded by an ‘s’ for silent, in which case no instructions or greeting is played (just a beep). Ifn the event no letter preceding letter is present, then only the instructions are played and not the greeting.
;
; This is the voicemail.conf basic example file for Opensource Asterisk server. Asterisk
; can be found on http://www.asterisk.org/
;
; More recent versions of this file can be found on:
; http://www.networks.pk
; Voicemail Configuration
[general]
; Default formats for writing Voicemail format=gsm|wav49|wav
; Who the e-mail notification should appear to come from
serveremail=voicemail@loligo.com
; Should the email contain the voicemail as an attachment attach=yes
; Maximum length of a voicemail message
;maxmessage=180
; Maximum length of greetings maxgreet=60
; Each mailbox is listed in the form =
,,
; if the e-mail is specified, a message will be sent when a message is
; received, to the given mailbox.
;
[default] 2203 => 9876,Syed Saud,syed@tele.pk 2204 => 5432,Froo Froo,nobody@fox-den.com No tag for this post.
Installing the TDM04B
Plug it into a spare PCI slot on your Asterisk server, just like any other expansion card. If you installed the Zaptel drivers when you installed Asterisk, you’re almost there. (If you didn’t, you need to re-install Asterisk and compile in Zaptel support.) The next step is to configure /etc/zaptel.conf. First make a backup copy of the original /etc/zaptel.conf: Read the rest of this entry »
Asterisk Zapata.conf configuration file is usually we use to configure Asterisk hardwares
i.e. FXO cards, FXS cards, T1/E1 cards. Below is an example zapata.conf file. You can edit it according to your needs. incase of any problem feel free to put comments.
Below is Open Source Asterisk PBX sip.conf configuration example file. Please edit it for your needs. If you need any help then feel free to put your comments.
No tag for this post.
